bug with moh no 180 or 183

[termy_pe]

Dont know if its a bug...

If you answer a did and you dial another number with the dial command...



exten => s,4,Dial(Local/00${CALLERID(dnid)}@outbound-allroutes|120)
exten => s,5,Hangup

Calling from another switch it will be silent and will last 120 seconds If you call from an extension it will play a fake ring for 120 seconds. You call from a extension from a another pbx, calling trough a switch then to the freepbx you will hear a fake ring.




Now the problem...



exten => s,4,Dial(Local/00${CALLERID(dnid)}@outbound-allroutes|120|m(default))
exten => s,5,Hangup

You hear the asterisk default music, and when calling from another switch, it will hangup after 60 seconds. Because, it does not send any ALERT_INFO. Like 180 Ringing or 183 Progress.

But if you call from extension from the same freepbx it will work, but not coming in from another freepbx going trough a switch server.

Is there a workaroung to make it ring and play moh in the same time, that way it will get a 180 and you could hear the musiconhold?

Please let me know, if its possible.

Thanks

 

[jeremywillden]

Can you clarify what you are trying to do? If I'm reading your dialplan code correctly, you are placing a call back to the caller's number, but adding a 00 to the beginning of the number received on the caller ID.

If that's what you are trying to do, use the callback module in FreePBX instead.

If you're trying to call out on another trunk (say to your cellphone) then you need to use the set callerID function, then do a dial to the destination number instead.

Examine (or post here) the stuff that's added to your /var/log/asterisk/full file when you call the DID and maybe I (we) can be more help. Cheers!

 

[termy_pe]

I use freepbx as a switch. So its like doing a call from trunk to trunk. I make a sip call to the PIAF. As you know, theres is no way you could do SIP to SIP. So I figure out a way. Make an incoming route with _XXNXXNXXXXXX

The XX is a prefix. So I know, when a call is sent its needs to be routed and correct the caller id.

So when I take the call, its the DID that becomes the CID. I put another prefix, and I send the call trough PIAF.

The outbound call, when its comes again is lets say. 0000|NXXNXXXXXX So its knows its a simple dial.

What I am trying to do, is for exemple, someone calls a 1800 number, I play a message please wait on the phone without picking the line up. And I call the other cell number or other pstn line number. But when I do this, PIAF or asterisk is not sending no alert message. So on most phone switch after 60 seconds its hangs up. Because I am calling with the MOH. But if I call with ringing, it could last 2 mins or whenever the switch has timeout setup for.

I need to send the ringing command, the 180 or 183 making progress, while the MOH is playing.

If you want try it. Take an incoming call, make it call an another phone play music. But remember you cannot answer the phone while the MOH is playing. So the person does not get billed, and you dont get billed for doing it for the 1800 number. until someone really picks up. I read from somewhere, that IBM customer support has a system like that. You even need to press 1 or 2. And when someone pick up the line its connected. I've been trough this for 1 month. I figure theres is no way. There is a bug in asterisk. Its works when you make the phone ring for over 60 sec. But with MOH its only work for 60 sec. Theres is something different between the two. The ring send the 180 back trough the sip. Not the MOH.

I will try to read the asterisk/log with SIP debug, and I guess I will post the result here. If my theory is right you will see the 180 on the ring and not on the moh.

I notice rogers a cell phone provider here, plays the ring+moh together, And I think it does send the 180, plays music + ring, for customer that pays extra. And of course I dont get charge from my cell phone until the person answer. That might be a solution, but is theres is no way you can add MOH+RING together?

Thank,

 

[termy_pe]

Hey. By googleing, I found someone this one that is looking for the same solution has me. As I read, when the dial command is used with the R, it send a 180 ring. When you put the m, it send a 183. A solution would be to write a patch, so send 180 with the m, or start with the r, and then send the moh with a 180 or 183. or something...

here is the link

http://www.mail-archive.com/[email protected]/msg222210.html

I think this problem is a bug or something, I dont know why it is not resolve, its seems that 183 does not do the trick.

Here is a interesting input from a solution:

Sipaddheader(180 Ringing) might do the trick.

If you are compiling your own asterisk, you could change chan_sip.c to
replace "183 Session Progress" with "180 Ringing" (line 3950 in my source)
but that might break something else.

Could that be done? How to you compile chan_sip.c?

I've never done that. And it looks like the best solution so far.


Thanks for any help...